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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5b4a29777dab756293048ea4de1a37dbafab5c79 / video
  1. a02d6e4 Skip encoding in fake VP8 encoder. by pbos@webrtc.org · 10 years ago
  2. a7651f8 Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
  3. d46f3eb Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 10 years ago
  4. 92cb201 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 10 years ago
  5. 921f06d Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  6. 42bd688 Extract RTP-header SSRC inline in Call. by pbos@webrtc.org · 10 years ago
  7. 8c71fbd Add test for VideoEncoder setup/teardown. by pbos@webrtc.org · 10 years ago
  8. 49e6306 Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  9. 9e2456f Fix data races related with traces in bitrate estimator test. by andresp@webrtc.org · 10 years ago
  10. b333ccd Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 10 years ago
  11. c311a0d Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  12. 6b32f7d Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  13. 59a8baa Adding pbos as video/ owner and removing persons never working with this folder. by mflodman@webrtc.org · 10 years ago
  14. b96302f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  15. 1539b4c Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  16. 97ce37c GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  17. 620f146 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  18. 212705c Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  19. ab25d49 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  20. 508e748 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  21. 1553688 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  22. 077efac Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
  23. 77cf8de Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  24. 6bb9535 Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. by wu@webrtc.org · 10 years ago
  25. 050a15e First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 10 years ago
  26. 1878192 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  27. da214b3 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  28. e62864e Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  29. 61ef0a3 Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  30. c23ed47 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  31. b4bc1a6 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  32. 1bc4fa6 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  33. 654bd9e Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
  34. 838c9da Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  35. 60f1422 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  36. 5e44f56 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  37. c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  38. 267637b Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  39. c298835 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  40. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  41. dbebc39 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
  42. 9d0f79f Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  43. 69b14d5 Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  44. 6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  45. 19ca463 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
  46. d8b4d0f Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  47. 4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  48. 1982636 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
  49. 6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  50. 765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  51. 0b11715 Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
  52. 539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  53. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  54. 9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  55. a183edc Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 10 years ago
  56. 292e7f6 Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 10 years ago
  57. 16c3dcc Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 10 years ago
  58. bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  59. f39df52 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  60. ebae8bb Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 10 years ago
  61. 8d3c410 Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 10 years ago
  62. f9a6ab0 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 10 years ago
  63. ca626eb Refactor rampup tests: by andresp@webrtc.org · 10 years ago
  64. 3c00b1c Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 10 years ago
  65. 15cf717 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
  66. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  67. 41da329 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 10 years ago
  68. c53e587 Replace labs with std::abs. by pbos@webrtc.org · 10 years ago
  69. af634a2 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
  70. f951dfc Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
  71. 697cd78 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 10 years ago
  72. f422ce1 Adding a link to issue by henrik.lundin@webrtc.org · 10 years ago
  73. b5c0d2e NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 10 years ago
  74. 4368a8f Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 10 years ago
  75. c63f18d Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 10 years ago
  76. 0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  77. deb5d53 Fix compilation errors under clang 3.5. by pbos@webrtc.org · 10 years ago
  78. 3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
  79. f2c28a0 Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
  80. fa7c4c4 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
  81. 4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
  82. fdb30d1 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 11 years ago
  83. 48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
  84. ffd4269 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
  85. c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  86. bb55b6d Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 11 years ago
  87. 64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  88. df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  89. c92ae91 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  90. ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  91. acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  92. cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  93. 2a4595a cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  94. b409d78 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
  95. f22f12a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  96. 4db3691 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  97. 620d9e5 Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  98. 4494516 Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  99. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  100. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago